OLD | NEW |
---|---|
1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
(...skipping 97 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
108 | 108 |
109 virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, | 109 virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, |
110 size_t incoming_packet_length) = 0; | 110 size_t incoming_packet_length) = 0; |
111 | 111 |
112 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; | 112 virtual void SetRemoteSSRC(uint32_t ssrc) = 0; |
113 | 113 |
114 // ************************************************************************** | 114 // ************************************************************************** |
115 // Sender | 115 // Sender |
116 // ************************************************************************** | 116 // ************************************************************************** |
117 | 117 |
118 // Sets MTU. | 118 // TODO(nisse): Use int or size_t? |
the sun
2016/12/21 10:33:02
size_t looks like the right call.
| |
119 // |size| - Max transfer unit in bytes, default is 1500. | 119 // Sets the maximum size of an RTP packet, including RTP headers. |
120 // Returns -1 on failure else 0. | 120 virtual void SetMaxRtpPacketSize(size_t size) = 0; |
121 virtual int32_t SetMaxTransferUnit(uint16_t size) = 0; | |
122 | 121 |
123 // Sets transport overhead. Default is IPv4 and UDP with no encryption. | 122 // Returns max payload length. |
124 // |tcp| - true for TCP false UDP. | |
125 // |ipv6| - true for IP version 6 false for version 4. | |
126 // |authentication_overhead| - number of bytes to leave for an authentication | |
127 // header. | |
128 // Returns -1 on failure else 0 | |
129 // TODO(michaelt): deprecate the function. | |
130 virtual int32_t SetTransportOverhead(bool tcp, | |
131 bool ipv6, | |
132 uint8_t authentication_overhead = 0) = 0; | |
133 | |
134 // Sets transport overhead per packet. | |
135 virtual void SetTransportOverhead(int transport_overhead_per_packet) = 0; | |
136 | |
137 // Returns max payload length, which is a combination of the configuration | |
138 // MaxTransferUnit and TransportOverhead. | |
139 // Does not account for RTP headers and FEC/ULP/RED overhead (when FEC is | 123 // Does not account for RTP headers and FEC/ULP/RED overhead (when FEC is |
140 // enabled). | 124 // enabled). |
141 virtual uint16_t MaxPayloadLength() const = 0; | 125 virtual size_t MaxPayloadSize() const = 0; |
142 | 126 |
143 // Returns max data payload length, which is a combination of the | 127 // Returns max data payload length, which is a combination of the |
144 // configuration MaxTransferUnit, headers and TransportOverhead. | 128 // configuration MaxTransferUnit, headers and TransportOverhead. |
michaelt
2016/12/20 14:11:42
nit: You removed this function. Right ?
nisse-webrtc
2017/01/09 16:02:24
Rephrased comment.
| |
145 // Takes into account RTP headers and FEC/ULP/RED overhead (when FEC is | 129 // Takes into account RTP headers and FEC/ULP/RED overhead (when FEC is |
146 // enabled). | 130 // enabled). |
147 virtual uint16_t MaxDataPayloadLength() const = 0; | 131 virtual size_t MaxRtpPacketSize() const = 0; |
148 | 132 |
149 // Sets codec name and payload type. Returns -1 on failure else 0. | 133 // Sets codec name and payload type. Returns -1 on failure else 0. |
150 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; | 134 virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; |
151 | 135 |
152 // Sets codec name and payload type. Return -1 on failure else 0. | 136 // Sets codec name and payload type. Return -1 on failure else 0. |
153 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; | 137 virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; |
154 | 138 |
155 virtual void RegisterVideoSendPayload(int payload_type, | 139 virtual void RegisterVideoSendPayload(int payload_type, |
156 const char* payload_name) = 0; | 140 const char* payload_name) = 0; |
157 | 141 |
(...skipping 324 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
482 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; | 466 virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
483 | 467 |
484 // Sends a request for a keyframe. | 468 // Sends a request for a keyframe. |
485 // Returns -1 on failure else 0. | 469 // Returns -1 on failure else 0. |
486 virtual int32_t RequestKeyFrame() = 0; | 470 virtual int32_t RequestKeyFrame() = 0; |
487 }; | 471 }; |
488 | 472 |
489 } // namespace webrtc | 473 } // namespace webrtc |
490 | 474 |
491 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ | 475 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
OLD | NEW |